- FreeSWITCH-users - FreeSWITCH Mailing Lists

Dangerous Demos Coming To ClueCon 2014!!
by Ken Rice 18 Jun '14

18 Jun '14

DANGEROUS DEMOS are coming to ClueCon! Our friends over at Truphone are at it again! Come to ClueCon to see the other Dangerous Demo, and while you are there show off your Dangerous Demo! Running for the first time in it¹s new and improved format, DANGEROUS DEMOS offers developers and hackers a chance to show off examples of their work in front of a live ClueCon audience! Competitors are invited to present a demonstration of something interesting that they have built. * Maximum demo duration will be 4 minutes * Bonus marks awarded for demos which > * Include FreeSWITCH > * Contain elements of uncertainty and/or danger > * Entertain and amuse the audience > * Present clear revenue generating potential > * Incorporate external parties/services > * Operate at multiple levels * Prizes: There will be at least THREE prizes (donated by Truphone), with the prospect of a FOURTH should special conditions be met. * Scoring will include ³Audience Participation² utilising a new realtime voting system (which would qualify as Dangerous Demo in it¹s own right) * Entries, containing a one or two sentence describing broadly what the demo may contain need to be sent to James Body (james.body(a)truphone.com) prior to the competition. Register for ClueCon today @ https://cluecon.com/

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questions about addons
by Jason Holden 18 Jun '14

18 Jun '14

Does Anyone have a recommendation for the following. 1. external speaker for paging with freeswitch, only transmit is required. 2. an application that may be loaded on a computer providing an operator full call routing capabilities. Thanks.

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DNS name as the public IP respond 403
by Shahzad Bhatti 18 Jun '14

18 Jun '14

Hi Brian, what do you mean by provider please explain a bit about provider is it about route provider or something else . Regards Shahzad Bhatti > ---------- Forwarded message ---------- > From: Brian West <brian(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Wed, 18 Jun 2014 08:21:44 -0500 > Subject: Re: [Freeswitch-users] DNS name as the public IP respond 403 > Again you need to talk to your provider they are the ones responding with > 403. > -- > Brian West > brian(a)freeswitch.org > > > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > On Jun 17, 2014, at 1:08 PM, Shahzad Bhatti <shahzad.bhatti(a)g-r-v.com> > wrote: > > here i post 2 sip traces first for successive registration with same >> credentials but using static ip >> > with headings as > > on line 40 > Sip Trace of Successive Call using Public IP > and error sip trace on line 288 > SIP TRACE OF USING DNS CALL FOR REGISTER (having problem) > > any further information is required to trace the issue i will happy to > send also > > regards > > Shahzad Bhatti > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

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No Ringback on few calls
by Nagalenoj H. 18 Jun '14

18 Jun '14

Hi, I am facing an inconsistent behavior in getting Ringback for calls. When I call to few numbers, I am not getting ringback tone. I did capture the packets and checked the difference between the call which I get RBT and failed case. I couldn't figure out what is stopping Freeswitch to provide '183 session progress' to the application after receiving '180 ringing' from the server on the other side. I checked the other server and it is sending the tone to FS, it is in FS machine the tone it not forwarded due to some reason. In Freeswitch log, there is no log for "Sending early media" in case of failure case. I've attached the packet capture for the calls. It has both working and non-working case. Please help me in debugging this issue.

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OpenSIPS Summit on Aug 4th.. come to cluecon early!
by Alex Goulis 18 Jun '14

18 Jun '14

The Windy City, Chi-Town, the Second City, the City of Big Shoulders, and the City That Works are just some of Chicago's many nicknames... *On August 4th, 2014 it will be the the "OpenSIPS" city!* If you've ever wondered how OpenSIPS can help you with your Freeswitch deployments, now's your chance. For all of you attending ClueCon, we'll be holding an OpenSIPS summit on Aug 4 running most of the day before ClueCon officially kicks off. *All ClueCon attendees receive a 25% discount by using the code "**cluecon2014".* Last year's summit was a tremendous success. Everyone that attended enjoyed interesting topics and discussions. Along with this announcement, we would like to announce next the two speakers for this years summit: *Steve Lorusso - Grasshopper - */"OpenSIPS in Production//"/ Engineers at Grasshopper recently deployed OpenSIPS as a production SBC for peering with multiple carriers. This session will cover what was done to setup, network, monitor, and operate OpenSIPS in an virtual HA configuration. Steve is VP of Engineering and Operations at Grasshopper, and oversees the technical direction, architecture, and implementation of software and services. He's got a track record of developing products from a 'startup' environment and guiding them to a mature business while building expert teams and processes. Before joining Grasshopper, Steve held technical management positions in Systems Engineering and Software Development. He was a day-1 core team member at Unisphere Networks, built SIP-based communications products for the Carrier marketplace, and then guided development through an acquisition (Siemens/Juniper) and subsequent global expansion into many countries. Steve is a seasoned engineering veteran with tier-1 Carrier and large Enterprise experience. He combines technical, business, and management experience with a passion for building innovative products. Steve has a number of patent applications relating to IP communications and redundant systems design. He holds a BSEE from Northeastern University and an MSEE degree from Boston University. *James Pearson - ViciDial - "*/Creating a routing and billing platform: How hard could it be?"/ Design and implementation of OpenSIPS into a billing and routing platform for a multi-tenanted high-capacity hosted call center. Topics include looking at what was required, design challenges and compromises, what wasn't known to be unknown, and other random craziness found along the way. James Pearson is a founder of The ViciDial Group in 2008 as well as the Director of Hosting for ViciHost.com. Currently maintains the OpenSIPS package on the OpenSuSE Build Service as well as contributing to other OBS projects. Experience with VoIP began in 2005 with Asterisk v.1.2 implementing customized PBXs for small business' as well as general network environment monitoring and design. Other areas of interest include networking, system automation, and new challenges You can get more information about the summit at: http://www.opensips.org/Community/Summit-2014Chicago Registration: http://www.opensips.org/Community/Summit-Registration Last year's summit highlights: http://www.opensips.org/Community/Summit-2013Chicago We look forward to seeing you! The OpenSIPS Team!

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DTMF I send are not recognized.
by Neo Haux 18 Jun '14

18 Jun '14

Hi all, I have an issue to send DTMF to an external extension (goes through my PSTN) My actual config is: |--|FXS-phone FS------LAN-----CISCO SPA3000--| |--| FXO-to-pstn Either I make a call from FXS of a soft SIP phone in my LAN the DTMF I send are not recognized at all by the number I call. When I send the DTMF I see in the console: freeswitch@internal> 2014-06-17 15:19:59.846809 [DEBUG] switch_rtp.c:5788 RTP RECV DTMF 1:1600 2014-06-17 15:19:59.846809 [DEBUG] switch_channel.c:486 RECV DTMF 1:1600 2014-06-17 15:19:59.846809 [DEBUG] switch_ivr_bridge.c:470 Send signal sofia/internal/5142252398@192.168.0.102:5061 [BREAK] 2014-06-17 15:19:59.846809 [DEBUG] switch_rtp.c:4314 Send start packet for [1] ts=1562418126 dur=160/160/1600 seq=48292 lw=1562418126 2014-06-17 15:19:59.866877 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=1562418126 dur=320/320/1600 seq=48293 lw=1562418286 2014-06-17 15:19:59.886848 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=1562418126 dur=480/480/1600 seq=48294 lw=1562418446 2014-06-17 15:19:59.906840 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=1562418126 dur=640/640/1600 seq=48295 lw=1562418606 2014-06-17 15:19:59.926964 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=1562418126 dur=800/800/1600 seq=48296 lw=1562418766 2014-06-17 15:19:59.946846 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=1562418126 dur=960/960/1600 seq=48297 lw=1562418926 2014-06-17 15:19:59.966838 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=1562418126 dur=1120/1120/1600 seq=48298 lw=1562419086 2014-06-17 15:19:59.986853 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=1562418126 dur=1280/1280/1600 seq=48299 lw=1562419246 2014-06-17 15:20:00.006849 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=1562418126 dur=1440/1440/1600 seq=48300 lw=1562419406 2014-06-17 15:20:00.026839 [DEBUG] switch_rtp.c:4213 Send end packet for [1] ts=1562418126 dur=1600/1600/1600 seq=48301 lw=1562419406 2014-06-17 15:20:00.026839 [DEBUG] switch_rtp.c:4213 Send end packet for [1] ts=1562418126 dur=1600/1600/1600 seq=48302 lw=1562419406 2014-06-17 15:20:00.046848 [DEBUG] switch_rtp.c:4213 Send end packet for [1] ts=1562418126 dur=1600/1600/1600 seq=48303 lw=1562419406 freeswitch@internal> freeswitch@internal> freeswitch@internal> freeswitch@internal> freeswitch@internal> freeswitch@internal> freeswitch@internal> freeswitch@internal> 2014-06-17 15:20:04.926811 [DEBUG] switch_rtp.c:5788 RTP RECV DTMF 1:1600 2014-06-17 15:20:04.926811 [DEBUG] switch_channel.c:486 RECV DTMF 1:1600 2014-06-17 15:20:04.926811 [DEBUG] switch_ivr_bridge.c:470 Send signal sofia/internal/5142252398@192.168.0.102:5061 [BREAK] 2014-06-17 15:20:04.926811 [DEBUG] switch_rtp.c:4314 Send start packet for [1] ts=1562459406 dur=160/160/1600 seq=48541 lw=1562459406 2014-06-17 15:20:04.946882 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=1562459406 dur=320/320/1600 seq=48542 lw=1562459566 2014-06-17 15:20:04.966839 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=1562459406 dur=480/480/1600 seq=48543 lw=1562459726 2014-06-17 15:20:04.986837 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=1562459406 dur=640/640/1600 seq=48544 lw=1562459886 2014-06-17 15:20:05.006847 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=1562459406 dur=800/800/1600 seq=48545 lw=1562460046 2014-06-17 15:20:05.026846 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=1562459406 dur=960/960/1600 seq=48546 lw=1562460206 2014-06-17 15:20:05.046839 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=1562459406 dur=1120/1120/1600 seq=48547 lw=1562460366 2014-06-17 15:20:05.066839 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=1562459406 dur=1280/1280/1600 seq=48548 lw=1562460526 2014-06-17 15:20:05.086839 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=1562459406 dur=1440/1440/1600 seq=48549 lw=1562460686 2014-06-17 15:20:05.106839 [DEBUG] switch_rtp.c:4213 Send end packet for [1] ts=1562459406 dur=1600/1600/1600 seq=48550 lw=1562460686 2014-06-17 15:20:05.106839 [DEBUG] switch_rtp.c:4213 Send end packet for [1] ts=1562459406 dur=1600/1600/1600 seq=48551 lw=1562460686 2014-06-17 15:20:05.126850 [DEBUG] switch_rtp.c:4213 Send end packet for [1] ts=1562459406 dur=1600/1600/1600 seq=48552 lw=1562460686 freeswitch@internal> freeswitch@internal> freeswitch@internal> When I receive a call and the caller click on digits (send me DTMF) I see this in the console: freeswitch@internal> 2014-06-17 15:18:06.466803 [DEBUG] switch_rtp.c:5788 RTP RECV DTMF 1:1604 2014-06-17 15:18:06.466803 [DEBUG] switch_channel.c:486 RECV DTMF 1:1604 2014-06-17 15:18:06.466803 [DEBUG] switch_ivr_bridge.c:470 Send signal sofia/internal/sip:1000@192.168.0.102:5060 [BREAK] 2014-06-17 15:18:06.486822 [DEBUG] switch_rtp.c:4314 Send start packet for [1] ts=407942050 dur=240/240/1604 seq=40146 lw=407942050 2014-06-17 15:18:06.506818 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=407942050 dur=480/480/1604 seq=40147 lw=407942290 2014-06-17 15:18:06.546821 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=407942050 dur=720/720/1604 seq=40148 lw=407942530 2014-06-17 15:18:06.566821 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=407942050 dur=960/960/1604 seq=40149 lw=407942770 2014-06-17 15:18:06.606824 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=407942050 dur=1200/1200/1604 seq=40150 lw=407943010 2014-06-17 15:18:06.626820 [DEBUG] switch_rtp.c:4213 Send middle packet for [1] ts=407942050 dur=1440/1440/1604 seq=40151 lw=407943250 2014-06-17 15:18:06.666823 [DEBUG] switch_rtp.c:4213 Send end packet for [1] ts=407942050 dur=1680/1680/1604 seq=40152 lw=407943250 2014-06-17 15:18:06.666823 [DEBUG] switch_rtp.c:4213 Send end packet for [1] ts=407942050 dur=1680/1680/1604 seq=40153 lw=407943250 2014-06-17 15:18:06.666823 [DEBUG] switch_rtp.c:4213 Send end packet for [1] ts=407942050 dur=1680/1680/1604 seq=40154 lw=407943250 I have no isea where to look. Is it a config to change within FS or SPA3000 ? What type of conf should I check ? In the past I had a Digium card+ asterisk and I had never had an issue with dtmf since the default config did work. I am new to FS so any idea is welcome. Thanks

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DNS name as the public IP respond 403
by Shahzad Bhatti 18 Jun '14

18 Jun '14

> > here i post 2 sip traces first for successive registration with same > credentials but using static ip > with headings as on line 40 Sip Trace of Successive Call using Public IP and error sip trace on line 288 SIP TRACE OF USING DNS CALL FOR REGISTER (having problem) any further information is required to trace the issue i will happy to send also regards Shahzad Bhatti ---------- Forwarded message ---------- > From: Brian West <brian(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 17 Jun 2014 12:59:02 -0500 > Subject: Re: [Freeswitch-users] DNS name as the public IP respond 403 > Raise the issue with your provider 403 usually means your credentials are > invalid. > > -- > Brian West > brian(a)freeswitch.org > > > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > On Jun 17, 2014, at 4:57 AM, Shahzad Bhatti <shahzad.bhatti(a)g-r-v.com> > wrote: > > Output of sofia status profile and sip trace log is here > > http://pastebin.freeswitch.com/22750 > > Regards > > Shahzad Bhatti > > ---------- Forwarded message ---------- >> From: Brian West <brian(a)freeswitch.org> >> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> Cc: >> Date: Mon, 16 Jun 2014 13:41:44 -0500 >> Subject: Re: [Freeswitch-users] DNS name as the public IP respond 403 >> Please pastebin a sip trace of this. And the output of 'sofia status >> profile internal' or what ever the profile name is. >> -- >> Brian West >> brian(a)freeswitch.org >> >> <whmcslogo.png> >> >> Twitter: @FreeSWITCH , @briankwest >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >> iNUM: +883 5100 1420 9001 >> ISN: 410*543 >> Skype:briankwest >> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >> >> >> On Jun 16, 2014, at 12:47 PM, Shahzad Bhatti <shahzad.bhatti(a)g-r-v.com> >> wrote: >> >> Hi, i am trying to use DNS Name and for that i start my Freeswitch Using >> -nonat and in sip profile create a new profile with the following changes >> >> <param name="sip-port" value="8080"/> >> >> and >> >> <param name="ext-rtp-ip" value="host:sip.mydomain-name.com"/> >> >> rest conf is same as internal and when i try to register i got 403 >> response while on internal profile it works fine >> i am using >> FreeSWITCH Version 1.5.8b+git~20131202T224806Z~e127ca7d40~64bit (git >> e127ca7 2013-12-02 22:48:06Z 64bit) >> >> on centos 64-bit >> >> reply me what else i am missing >> >> regards >> >> Shahzad Bhatti >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

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Fwd: Recording in flac format ?
by julien terrasson 18 Jun '14

18 Jun '14

Hi, I'm using freeswitch version 1.4.5 I would like to record a session in .flac format. I assume this could be somehow possible since FS rely on libsndfile-1.0.26pre5 and libsndfile claims flac support since v1.0.18. I tried to compiled FS with instruction from description of https://jira.freeswitch.org/browse/FS-1197 : Apt-get install libflac-dev and "./configure --enable-flac".. However that didn't help, flac extension is still not usable on FS recording app. Does anyone have an idea how to have flac format enabled ? Regards, J.Terrasson

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 96, Issue 137
by Amin Dhamani Askhanuman 17 Jun '14

17 Jun '14

Yes its problem on inbound side. But where to set this parameter? in vars.xml? On Tue, Jun 17, 2014 at 1:40 AM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: Voice Mail issue (Brian West) > > > ---------- Forwarded message ---------- > From: Brian West <brian(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Mon, 16 Jun 2014 13:40:15 -0500 > Subject: Re: [Freeswitch-users] Voice Mail issue > Answer, then do playback silence_stream://1500, that should clear it, > Sonus on the inbound side? > > -- > Brian West > brian(a)freeswitch.org > > > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > > > On Jun 16, 2014, at 9:55 AM, Amin Dhamani Askhanuman <amin(a)askhanuman.com> > wrote: > > Hi, > > Currently we are facing issue with voicemail. Sound quality is not good. > Even caller can't hear the greeting properly. Can you guide how to resolve > it. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Best regards, Amin Dhamani Amin Dhamani | System Administrator | +66 911167159 | Krits Tower, 1032/1-5, 14 Rama 4, Bangkok 10120 *www.askhanuman.co.th <http://www.askhanuman.co.th/>* | www.facebook.com/AskHanuman | amin(a)askhanuman.com This communication may contain information that is legally privileged, confidential or exempt from disclosure. If you are not the intended recipient, please note that any dissemination, distribution, or copying of this communication is strictly prohibited. Anyone who receives this message in error should notify the sender immediately by telephone or by return e-mail and delete this communication entirely from his or her computer. This communication may contain information that is legally privileged, confidential or exempt from disclosure. If you are not the intended recipient, please note that any dissemination, distribution, or copying of this communication is strictly prohibited. Anyone who receives this message in error should notify the sender immediately by telephone or by return e-mail and delete this communication entirely from his or her computer. ______________________________________________________________________________ -- This communication may contain information that is legally privileged, confidential or exempt from disclosure. If you are not the intended recipient, please note that any dissemination, distribution, or copying of this communication is strictly prohibited. Anyone who receives this message in error should notify the sender immediately by telephone or by return e-mail and delete this communication entirely from his or her computer. ______________________________________________________________________________

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DNS name as the public IP respond 403
by Shahzad Bhatti 17 Jun '14

17 Jun '14

Output of sofia status profile and sip trace log is here http://pastebin.freeswitch.com/22750 Regards Shahzad Bhatti ---------- Forwarded message ---------- > From: Brian West <brian(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Mon, 16 Jun 2014 13:41:44 -0500 > Subject: Re: [Freeswitch-users] DNS name as the public IP respond 403 > Please pastebin a sip trace of this. And the output of 'sofia status > profile internal' or what ever the profile name is. > -- > Brian West > brian(a)freeswitch.org > > > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > On Jun 16, 2014, at 12:47 PM, Shahzad Bhatti <shahzad.bhatti(a)g-r-v.com> > wrote: > > Hi, i am trying to use DNS Name and for that i start my Freeswitch Using > -nonat and in sip profile create a new profile with the following changes > > <param name="sip-port" value="8080"/> > > and > > <param name="ext-rtp-ip" value="host:sip.mydomain-name.com"/> > > rest conf is same as internal and when i try to register i got 403 > response while on internal profile it works fine > i am using > FreeSWITCH Version 1.5.8b+git~20131202T224806Z~e127ca7d40~64bit (git > e127ca7 2013-12-02 22:48:06Z 64bit) > > on centos 64-bit > > reply me what else i am missing > > regards > > Shahzad Bhatti > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

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